And also, what a2b version you are use? If you are use 1.7 then all config is in DB, if 1.3(4) all config in a2billing.conf
-- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: [email protected] www.eif-it.com Vardan Harutyunyan wrote: > I send you my a2b config for whole sale > > use_dnid = YES - this is the main option that you must use > > You can call this config like so: > DeadAGI(a2billing.php|3) > > I hope this will be help you. > > [agi-conf3] > > ; the debug level > ; 0=none, 1=low, 2=normal, 3=all > debug = 0 > > ; Asterisk Version Information > ; 1_1,1_2,1_4 By Default it will take 1_2 or higher > asterisk_version = 1_4 > > ; Manage the answer on the call > answer_call = NO > > ; Play audio - this will disable all stream file but not the Get Data > ; for wholesale ensure that the authentication works and than number_try = 1 > play_audio = NO > > ; play the goodbye message when the user has finished. > say_goodbye = NO > > ; enable the menu to choose the language > ; press 1 for English, pulsa 2 para el espaУБol, Pressez 3 pour FranУЇais > play_menulanguage = NO > > > ; force the use of a language, if you dont want to use it leave the > option empty > ; Values : ES, EN, FR, etc... (according to the audio you have installed) > force_language = > > ; Introduction prompt : to specify an additional prompt to play at the > beginning of the application > intro_prompt = > > ; Minimum amount of credit to use the application > min_credit_2call = 0 > > ; this is the minimum duration in seconds of a call in order to be billed > ; any call with a length less than min_duration_2bill will have a 0 cost > ; useful not to charge callers for system errors when a call was > answered but it actually didn't connect > min_duration_2bill = 0 > > ; if user doesn't have enough credit to call a destination, prompt him > to enter another cardnumber > notenoughcredit_cardnumber = NO > > ; if notenoughcredit_cardnumber = YES then assign the CallerID to > the new cardnumber > notenoughcredit_assign_newcardnumber_cid = NO > > > ; if YES it will use the DNID and try to dial out, without asking for > the phonenumber to call > ; value : YES, NO > use_dnid = YES > > ; list the dnid on which you want to avoid the use of the previous > option "use_dnid" > no_auth_dnid = 2400,2300 > > ; number of times the user can dial different number > number_try = 1 > > ; this will force to select a specific call plan by the Rate Engine > force_callplan_id = > > ; Play the balance to the user after the authentication (values : yes - no) > say_balance_after_auth = NO > > ; Play the balance to the user after the call (values : yes - no) > say_balance_after_call = NO > > ; Play the initial cost of the route (values : yes - no) > say_rateinitial = NO > > ; Play the amount of time that the user can call (values : yes - no) > say_timetocall = NO > > > ; enable the setup of the callerID number before the outbound is made, > by default the user callerID value will be use > auto_setcallerid = NO > > ; If auto_setcallerid is enabled, the value of force_callerid will be > set as CallerID > force_callerid = > > ; If force_callerid is not set, then the following option ensures that > CID is set to one of the card's configured caller IDs or blank if none > available. > ; NO - disable this feature, caller ID can be anything. > ; CID - Caller ID must be one of the customers caller IDs > ; DID - Caller ID must be one of the customers DID nos. > ; BOTH - Caller ID must be one of the above two items. > cid_sanitize = NO > > > ; enable the callerid authentication > ; if this option is active the CC system will check the CID of caller > cid_enable = NO > > ; if the CID does not exist, then the caller will be prompt to enter his > cardnumber > cid_askpincode_ifnot_callerid = NO > > ; if the callerID authentication is enable and the authentication fails > then the user will be prompt to enter his cardnumber > ; this option will bound the cardnumber entered to the current callerID > so that next call will be directly authenticate > cid_auto_assign_card_to_cid = NO > > ; if the callerID is captured on a2billing, this option will create > automatically a new card and add the callerID to it > cid_auto_create_card = NO > > ; set the length of the card that will be auto create (ie, 10) > cid_auto_create_card_len = 10 > > ; If cid_auto_create_card has been set to YES, the following options > will define with which configuration we will create the card > ; > ; billing type of the new card > ; ( value : POSTPAY or PREPAY) > cid_auto_create_card_typepaid = POSTPAY > > ; amount of credit of the new card > cid_auto_create_card_credit = 0 > > ; if postpay, define the credit limit for the card > cid_auto_create_card_credit_limit = 1000 > > ; the tariffgroup to use for the new card (this is the ID that you can > find on the admin web interface) > cid_auto_create_card_tariffgroup = 6 > > ; to check callerID over the cardnumber authentication (to guard against > spoofing) > callerid_authentication_over_cardnumber = NO > > ; enable the option to call sip/iax friend for free (values : YES - NO) > sip_iax_friends = no > > ; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed > digits to call a pstn number > ; values : number > sip_iax_pstn_direct_call_prefix = 555 > > ; this will enable a prompt to enter your destination number. > ; if number start by sip_iax_pstn_direct_call_prefix we do directly a > sip iax call, if not we do a normal call > sip_iax_pstn_direct_call = NO > > ; enable the option to refill card with voucher in IVR (values : YES - NO) > ivr_voucher = NO > > ; if ivr_voucher is active, you can define a prefix for the voucher > number to refill your card > ; values : number - don't forget to change > prepaid-refill_card_with_voucher audio accordingly > ivr_voucher_prefix = 8 > > ; When the user credit are below the minimum credit to call min_credit > ; jump directly to the voucher IVR menu (values: YES - NO) > jump_voucher_if_min_credit = NO > > ; Extracharge DIDs, multiple numbers and fees must be separated by comma > ; extracharge_did = 1800XXXXXXX,1888XXXXXXX > extracharge_did = > ;extracharge_fee = 0.02,0.03 > extracharge_fee = > > ; List the prefixes that will be stripped off if the call plan requires it > international_prefixes = 9999999999999 > > ; More information about the Dial : > http://voip-info.org/wiki-Asterisk+cmd+dial > ; 30 : The timeout parameter is optional. If not specifed, the > Dial command will wait indefinitely, exiting only when the originating > channel hangs up, or all the dialed channels return a busy or error > condition. Otherwise it specifies a maximum time, in seconds, that the > Dial command is to wait for a channel to answer. > ; H: Allow the caller to hang up by dialing * > ; r: Generate a ringing tone for the calling party > ; g: When the called party hangs up, exit to execute more commands > in the current context. (new in 1.4) > ; i: Asterisk will ignore any forwarding (302 Redirect) requests > received. Essential for DID usage to prevent fraud. (new in 1.4) Useful > if you are ringing a group of people and one person has set their phone > to forwarded direct to voicemail on their cell or something which > normally prevents any of the other phones from ringing. > ; R: Indicate ringing to the calling party when the called party > indicates ringing, pass no audio until answered. > ; m: Provide Music on Hold to the calling party until the called > channel answers. > ; L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are > left, repeated every 'z' ms) > ; %timeout% tag is replaced by the > calculated timeout according the credit& destination rate! > > ;dialcommand_param = "|60|HRgrL(%timeout%:61000:30000)" > ;dialcommand_param = "|60|gL(%timeout%)" > dialcommand_param = "|60|gS(%timeout%)" > ;dialcommand_param = "|60|g" > > ; by default (3600000 = 1HOUR MAX CALL) > dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)" > > ; Define the order to make the outbound call > ; YES -> SIP/dialedphonenum...@gateway_ip - NO > SIP/gateway_ip/dialedphonenumber > ; Both should work exactly the same but i experimented one case when > gateway was supporting dialedphonenum...@gateway_ip > ; So in case of trouble, try it out > switchdialcommand = yes > > ; failover recursive search - define how many time we want to authorize > the research of the failover trunk when a call fails (value : 0 - 20) > failover_recursive_limit = 2 > > ; For free calls, limit the duration: amount in seconds > maxtime_tocall_negatif_free_route = 5400 > > ; Send a reminder email to the user when they are under min_credit_2call > send_reminder = NO > > ; enable to monitor the call (to record all the conversations) > ; value : YES - NO > record_call = NO > > ; format of the recorded monitor file > monitor_formatfile = gsm > > ; Force to play the balance to the caller in a predefined currency, to > use the currency set for by the customer leave this field empty > agi_force_currency = > > ; CURRENCY SECTION > ; Define all the audio (without file extensions) that you want to play > according to currency (use , to separate, ie > "usd:prepaid-dollar,mxn:pesos,eur:Euro,all:credit") > currency_association = usd:dollars,mxn:pesos,eur:euros,all:credit > > ; Please enter the file name you want to play when we prompt the calling > party to enter the destination number > ; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011 > file_conf_enter_destination = prepaid-enter-dest > > ; Please enter the file name you want to play when we prompt the calling > party to choose the prefered language > ; file_conf_enter_menulang = prepaid-menulang > file_conf_enter_menulang = prepaid-menulang2 > > ; Define if you want to bill the 1st leg on callback even if the call is > not connected to the destination > callback_bill_1stleg_ifcall_notconnected = YES > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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